Professional Documents
Culture Documents
Sampling
Experiment No.1
Digital Communications Laboratory I
Third Stage
Morning Study
Group (B)
➢ Objective
The objective of this experiment is to understand concepts and observe the effects of
periodically sampling a continuous signal at different sampling rates.
➢ Theory
In order to store, transmit or process analog signals using digital hardware, we must first
convert them into discrete-time signals by sampling. The processed discrete-time signal is
usually converted back to analog form by interpolation, resulting in a reconstructed analog
signal 𝑋𝑟 (𝑡). Sampling and Reconstruction process is shown in figure (1). The sampler reads
the values of the analog signal 𝑋𝑎 (𝑡) at equally spaced sampling instants. The time interval
𝑇𝑠 between adjacent samples is known as the sampling period (or sampling interval). The
1
sampling rate, measured in samples per second, is 𝑓𝑠 =
𝑇𝑠
The uniform sampling theorem states that a band limited signal having no spectral
components above 𝑓𝑚 hertz can be determined uniquely by values sampled at uniform
intervals of:
1
𝑇𝑠 ≤
2𝑓𝑚
1
The upper limit on Ts can be expressed in terms of sampling rate, denoted 𝑓𝑠 = . The
𝑇𝑠
restriction, stated in term of the sampling rate, is known as the Nyquist criterion.
Digital Communications Laboratory I Experiment No.1
The sampling rate is also called Nyquist rate. The allow Nyquist criterion is a theoretically
sufficient condition to allow an analog signal to be reconstructed completely from a set of a
uniformly spaced discrete-time samples.
➢ Procedure
1. Implement the block diagram shown in figure (3) using MATLAB Simulink.
Figure (3)
𝑟𝑎𝑑
2. Set the Signal Generator 1 𝑉𝑃 and a radian frequency of 1 .
𝑠𝑒𝑐
Digital Communications Laboratory I Experiment No.1
3. Set the block parameters of the trigger as: Pulse type: Time based, Amplitude = 1,
period = 0.5 sec, pulse width = 50%, and phase delay = 0.
4. Set the block parameters of the analog filter as filter order = 2, filter edge frequency =
𝑟𝑎𝑑
1.1 .
𝑠𝑒𝑐
6. Connect the block diagram shown in Figure (4) using MATLAB Simulink.
Figure (4)
𝑟𝑎𝑑
7. Set the Signal Generator to 1 𝑉𝑃 and a radian frequency of 1 .
𝑠𝑒𝑐
8. Set the block parameters of the trigger as: Pulse type: Time based, Amplitude = 1,
period = 0.9 sec, pulse width = 50%, and phase delay = 0.
9. Set the block parameters of the analog filter as filter order = 2, filter edge frequency =
𝑟𝑎𝑑
1.1 .
𝑠𝑒𝑐
➢ Discussion
1. What are the advantages of sampling analog signals?
❖ The sampling process is the first stage from the stages of converting an analog
signal (e.g., a human voice signal) into a digital signal.
2. What are the following terms mean: Instantaneous Sampling, Sampling and Flat-Top
Sampling?
❖ Instantaneous Sampling, it one of the types of sampling theory of an analog
information signal 𝑚(𝑡) that is multiplied by a unit impulse train 𝑃(𝑡) yields, for
the purpose of converting it into a digital signal, and considered the ideal form of
sampling.
Where:
𝑃(𝑡) = ∑+∞
𝑛=−∞ 𝛿(𝑡 − 𝑛𝑇𝑠 )
𝑚𝑠 (𝑡) = 𝑚(𝑡)𝑃(𝑡)
+∞
❖ Sampling, it the process of cutting the analog signal in the time domain to the
purpose of sending more than one signal through the channel with the possibility
of its recovery, and it states: The original signal can be restored from the sampled
signal if sampling frequency (𝑓𝑠 ) more than or equal to the double of frequency of
information signal (2𝑓𝑚 )
❖ Flat-Top Sampling, it the most common practical sampling methods that also called
the sample-and-hold (S/H) where the amplitude of sampled signal will be the equal
to the amplitude of information signal but it will be flat and constant.
Digital Communications Laboratory I Experiment No.1
4. Explain the reason for making the sampling frequency greater than or equal to (2B) Hz,
where B is the bandwidth of the analog signal.
❖ To be able to retrieve the information signal from the sampled signal and avoid the
occurrence "aliasing" that occur as a result interference of adjacent frequency
components with each other in the frequency domain sampling spectrum, and to
that, we need to keep the minimum sampling frequency which is twice the
bandwidth of the information signal.